The Internet is increasingly being used for the transmission of high volumes of multimedia data generated from applications like video on demand, video telephony and videoconferencing. Reliable transmission of video over packet switched telecommunication networks such as the Internet requires stringent bandwidth demand as well as low delay and packet loss requirements. Currently, the Internet does not include any particular function in the network layer to support these requirements. Instead, the service quality functions are provided for at the application layer of the protocol stack or in the Transmission Control Protocol (TCP) found at the stack's transport layer.
TCP incorporates algorithms that are implemented on top of the network layer to control network congestion by adjusting the transmission rate based on the packet loss ratio and the round-trip time (RTT) experienced by data during transmission. To regulate network congestion it is therefore important to have an accurate estimation of the round-trip time. Other protocols that regulate real-time network traffic also make use of RTT estimates.
The round-trip time is the time interval that elapses between the sending of a packet and the receipt of its acknowledgement. Individual RTT samples can be obtained by a protocol that sends packets from the source to destination and tracks their delivery. Although it is fairly straightforward to design such a protocol, the accuracy of individual samples is not a very good representation of the actual RTT due to various noise sources such as the difference in synchronization between the sender's and the receiver's clocks. Rather than using the sampled value directly, most traffic controllers use a prediction of the future values, as the predicted values give better results than the sampled values.
Prediction of round-trip time is complicated by the noisy and nonlinear nature of the sampled values. Currently, TCP predicts future round trip times by sampling the behaviour of packets sent over a connection and averaging those samples into a “smoothed” round-trip time estimate value using a digital filter. The digital filter used in TCP belongs to the family of Fading Memory Polynomial (FMP) filters of degree zero. FMP filters produce an estimated value based on a weighted sequence of prior values and prior estimated values. FMP filters are not self-initialising, and accordingly the values predicted at the start-up of TCP connections are drastically affected by the initial input to the filter. The start-up transient behaviour of the TCP filter makes the initial estimates produced by the TCP filter inaccurate.
In K. Jacobsson, H. Hjalmarsson, N. Moller and K. H. Johanson, “Round-Trip Time Estimation in Communication Networks Using Adaptive Kalman Filtering,” KTH—Royal Institute of Technology, 2004, an adaptive Kalman filter is introduced with a Cumulative Sum (CUSUM) change detection scheme to detect sudden changes in the average RTT. Although the filter corrects the start-up problem of the TCP filter it introduces other challenges, such as the online determination of drift and the possibility of misdetection when a change on the average RTT occurs.